1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
|
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
#include <stdio.h>
#include <string.h>
#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#include "webrtc/modules/audio_coding/main/test/RTPFile.h"
#include "webrtc/typedefs.h"
namespace webrtc {
#define MAX_INCOMING_PAYLOAD 8096
// TestPacketization callback which writes the encoded payloads to file
class TestPacketization : public AudioPacketizationCallback {
public:
TestPacketization(RTPStream *rtpStream, uint16_t frequency);
~TestPacketization();
int32_t SendData(const FrameType frameType,
const uint8_t payloadType,
const uint32_t timeStamp,
const uint8_t* payloadData,
const size_t payloadSize,
const RTPFragmentationHeader* fragmentation) override;
private:
static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
int16_t seqNo, uint32_t timeStamp, uint32_t ssrc);
RTPStream* _rtpStream;
int32_t _frequency;
int16_t _seqNo;
};
class Sender {
public:
Sender();
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
std::string in_file_name, int sample_rate, int channels);
void Teardown();
void Run();
bool Add10MsData();
//for auto_test and logging
uint8_t testMode;
uint8_t codeId;
protected:
AudioCodingModule* _acm;
private:
PCMFile _pcmFile;
AudioFrame _audioFrame;
TestPacketization* _packetization;
};
class Receiver {
public:
Receiver();
virtual ~Receiver() {};
void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
std::string out_file_name, int channels);
void Teardown();
void Run();
virtual bool IncomingPacket();
bool PlayoutData();
//for auto_test and logging
uint8_t codeId;
uint8_t testMode;
private:
PCMFile _pcmFile;
int16_t* _playoutBuffer;
uint16_t _playoutLengthSmpls;
int32_t _frequency;
bool _firstTime;
protected:
AudioCodingModule* _acm;
uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];
RTPStream* _rtpStream;
WebRtcRTPHeader _rtpInfo;
size_t _realPayloadSizeBytes;
size_t _payloadSizeBytes;
uint32_t _nextTime;
};
class EncodeDecodeTest : public ACMTest {
public:
EncodeDecodeTest();
explicit EncodeDecodeTest(int testMode);
void Perform() override;
uint16_t _playoutFreq;
uint8_t _testMode;
private:
std::string EncodeToFile(int fileType,
int codeId,
int* codePars,
int testMode);
protected:
Sender _sender;
Receiver _receiver;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
|
